Cost savings through free or low-cost calls, especially over the internet.
Flexibility with use on computers, smartphones, and tablets.
Features like call hold, transfer, forwarding, SMS, chat, and conference calling.
Integration with CRMs, Microsoft Teams and other business tools.
Scalability to accommodate growth without significant infrastructure changes.
Tragofone supports HD voice & video calls with features like call transfer, voicemail, call recording, and 3-way conferencing for smooth professional communication.
Tragofone integrates with enterprise tools like HubSpot, noCRM, PortaSwitch, Microsoft Teams, NetSapiens, Zoom and more.
You can fully rebrand Tragofone, change its name, logo, colors, and UI to match your business. From interface language to call settings, everything is customizable for a branded experience.
Stay connected through one-to-one or group chat, SMS, and MMS. You can sync contacts, share files, and track availability with real-time presence, all from one place.
Using SIP integrations with Push Call Server (PCS), Tragofone ensures desktop and mobile devices receive call alerts in real time, even when the app is closed.
Tragofone simplifies large-scale deployments with SIP-based auto-provisioning. Admins can configure settings remotely via a centralized dashboard.
Tracks SIP-level metrics like session quality, network type, call abandonment, and device-specific logs to help businesses monitor and optimize call performance.
SIP-enabled contact syncing allows Tragofone to integrate with enterprise directories, LDAP, cloud contacts from Google and Microsoft, so users have unified access to their business contacts.
Enables SIP-based call management capabilities such as transferring calls, putting them on hold, and VoIP conferencing calls—crucial for business-grade telephony.
Tragofone protects your conversations using TLS, SRTP, DTLS, and secure WebSocket protocols.
Tragofone supports simultaneous multi-device login, syncing SIP sessions and call logs in real time across desktop and mobile.
Tragofone allows SIP rule-based forwarding of calls and supports emergency number routing as defined in enterprise SIP policies.
Determines whether the party that is being called is available to take it.
Defines the media streams to be used for the call.
Delivers push notifications to iOS devices using APNS and to Android devices using Google FCM.
Connects a call from one user to the other.
Running of the call session. For example: transferring and ending a call.
When a user agent launches, it has to register with a VoIP SIP server so it can be found by other users. This happens by using the SIP Register request message.
The VoIP call is transmitted between the user agents using RTP (real-time transport protocol), which is also used for sending audio and video data over IP networks.
When a user ends a call, a SIP softphone client Bye request is sent. The call can be ended from either side of the line. The termination is confirmed by the other user agent with a SIP 200 status code response.
Reduces infrastructure costs by replacing traditional phone lines with SIP trunks
Supports voice, video, messaging, and presence information in a single platform
Enables call transfer, voicemails, auto-attendants, conferencing, and CRM integration
Supports encryption protocols like TLS, SRTP and WebSocket
Adapts to emerging business communication technologies like WebRTC and IoT
Verify that your existing hardware, routers, and wireless network are fully capable of efficiently handling Quality of Service (QoS) settings and support a SIP-based environment.
Ensure your network has sufficient bandwidth to handle VoIP traffic alongside existing data usage. Upgrade to a VoIP phone system, or some traditional phones can be upgraded with (ATAs).
To support SIP calling, you may need to upgrade your existing PBX with SIP trunking. It is also advisable to switch to a cloud-based communication solution to keep the transition smooth and future-proofing.
Get quick answers to common questions about our VoIP softphone solutions and features